/**********
This library is free software; you can redistribute it and/or modify it under
the terms of the GNU Lesser General Public License as published by the
Free Software Foundation; either version 3 of the License, or (at your
option) any later version. (See <http://www.gnu.org/copyleft/lesser.html>.)

This library is distributed in the hope that it will be useful, but WITHOUT
ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS
FOR A PARTICULAR PURPOSE.  See the GNU Lesser General Public License for
more details.

You should have received a copy of the GNU Lesser General Public License
along with this library; if not, write to the Free Software Foundation, Inc.,
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
**********/
// "liveMedia"
// Copyright (c) 1996-2018 Live Networks, Inc.  All rights reserved.
// MPEG-1 or MPEG-2 Audio RTP Sources
// Implementation

#include "include/MPEG1or2AudioRTPSource.hh"

MPEG1or2AudioRTPSource *
MPEG1or2AudioRTPSource::createNew(UsageEnvironment &env,
                                  Groupsock *RTPgs,
                                  unsigned char rtpPayloadFormat,
                                  unsigned rtpTimestampFrequency) {
    return new MPEG1or2AudioRTPSource(env, RTPgs, rtpPayloadFormat,
                                      rtpTimestampFrequency);
}

MPEG1or2AudioRTPSource::MPEG1or2AudioRTPSource(UsageEnvironment &env,
                                               Groupsock *rtpGS,
                                               unsigned char rtpPayloadFormat,
                                               unsigned rtpTimestampFrequency)
        : MultiFramedRTPSource(env, rtpGS,
                               rtpPayloadFormat, rtpTimestampFrequency) {
}

MPEG1or2AudioRTPSource::~MPEG1or2AudioRTPSource() {
}

Boolean MPEG1or2AudioRTPSource
::processSpecialHeader(BufferedPacket *packet,
                       unsigned &resultSpecialHeaderSize) {
    // There's a 4-byte header indicating fragmentation.
    if (packet->dataSize() < 4) return False;

    // Note: This fragmentation header is actually useless to us, because
    // it doesn't tell us whether or not this RTP packet *ends* a
    // fragmented frame.  Thus, we can't use it to properly set
    // "fCurrentPacketCompletesFrame".  Instead, we assume that even
    // a partial audio frame will be usable to clients.

    resultSpecialHeaderSize = 4;
    return True;
}

char const *MPEG1or2AudioRTPSource::MIMEtype() const {
    return "audio/MPEG";
}

